As far as I can tell Asterisk recives media perfectly. For outgoing calls it looks something like this:<br><br> -- Executing [xxx@proxy:5] WaitExten("SIP/voiptrunk-00000083", "7") in new stack<br>DEBUG[28557]: rtp.c:1032 process_rfc2833: - RTP 2833 Event: 00000001 (len = 4)<br>
DEBUG[28557]: rtp.c:880 send_dtmf: Sending dtmf: 49 (1), at xx.xx.xxx.x<br><br>On incoming, as far as I can tell, Asterisk does not recieve anything. I just don't know why.<br><br>I have added exceptions in firewall and network to allow voip traffic, successfully allowing incoming and outgoing calls. Just no DTMF on incoming calls.<br>
<br>My tests consist of a regular landline, I dial a DID and successfully reach my asterisk box. Everything is fine until I come to user input. None is recognized. I get a "-User entered nothing" and timeout. <br>
<br><br><br><br><div class="gmail_quote">On Mon, Aug 23, 2010 at 8:07 AM, Miguel Molina <span dir="ltr"><<a href="mailto:mmolina@millenium.com.co">mmolina@millenium.com.co</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
El 20/08/10 16:14, Kathryn Jones escribió:<br>
<div class="im">> Thanks for all the help, but I still can't find what's wrong.<br>
><br>
> I enabled console => notice,warning,error,debug,dtmf like Miguel<br>
> suggested. The output is attached.<br>
><br>
> I noticed that the rtp.c session never starts, which as I understand<br>
> is what catches the dtmf tone, but I could not find how to start it :s.<br>
><br>
> The Answer() and waitExten(5,m) didn't fix my problem. I hope someone<br>
> can help me see the problem after looking at the attached console output.<br>
</div>The following line brought my attention:<br>
<br>
[Aug 20 16:50:04] DEBUG[5319]: channel.c:1882 __ast_answer: Didn't receive a media frame from SIP/xx.xx.xxx.xx-00000026 within 500 ms of answering. Continuing anyway<br>
<br>
<br>
<br>
Are your sure that RTP audio (media) is correctly received in asterisk?<br>
I suspect network or firewall problems. Also, you said that you were<br>
going to receive calls from the PSTN, but are you testing from a SIP<br>
endpoint?<br>
<br>
Regards,<br>
<div class="im"><br>
--<br>
Ing. Miguel Molina<br>
Grupo de Tecnología<br>
Millenium Phone Center<br>
<br>
<br>
</div>--<br>
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